No se escucha en los telefonos SIP los mensajes que se deben

**rbknauj** · 02-19-2005, 02:25 PM

No se escucha en los telefonos SIP los mensajes que se deben

a mi me paso una vez y cambie el dtmfmode y me funciono

**akrall** · 02-19-2005, 05:02 PM

No se escucha en los telefonos SIP los mensajes que se deben

a que lo cambiaste?

**Invitado** · 02-19-2005, 09:17 PM

No se escucha en los telefonos SIP los mensajes que se deben

Que es ya lo convertiste a gsm?

**akrall** · 02-19-2005, 09:19 PM

No se escucha en los telefonos SIP los mensajes que se deben

Estoy usando el demo de asterisk, asi que son los sonidos default.. esto es lo que dice el sip debug:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)

Asi que veo que si acuerdan en ilbc... pero, porque no escucho, que raro porque en el demo que hice hace un par de dias si se escuchaba y ahora no se que pedo..

**Invitado** · 02-19-2005, 09:23 PM

No se escucha en los telefonos SIP los mensajes que se deben

Primero explica el esenario que estas tratando de hacer?

**akrall** · 02-19-2005, 09:27 PM

No se escucha en los telefonos SIP los mensajes que se deben

Acabo de instalar RH9, puse todo lo que se debe en dependencias, baje el * y zaptel, compile todo, hice el make samples, modifico el sip.conf para que se de de alta un telefono, arranco asterisk... marco la extension 1000 que es la bienvenida del demo... y nada, esto e slo que dice asterisk:

-- Executing Goto("SIP/grandstream1-2bb7", "default|s|1") in new stack
-- Goto (default,s,1)
-- Executing Wait("SIP/grandstream1-2bb7", "1") in new stack
-- Executing Answer("SIP/grandstream1-2bb7", "") in new stack
-- Executing DigitTimeout("SIP/grandstream1-2bb7", "5") in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("SIP/grandstream1-2bb7", "10") in new stack
-- Set Response Timeout to 10
-- Executing BackGround("SIP/grandstream1-2bb7", "demo-congrats") in new stack
-- Playing 'demo-congrats' (language 'en')
== Spawn extension (default, s, 5) exited non-zero on 'SIP/grandstream1-2bb7'

contesta el asterisk pero no escucho nada en el telefono, la grabacion de bienvenida no suena.

**akrall** · 02-19-2005, 09:33 PM

No se escucha en los telefonos SIP los mensajes que se deben

Por cierto, en el log veo este error, no se si tenga algo que ver:

Feb 19 21:31:49 WARNING[26971]: Unable to open /dev/dsp: No such device
y este:
Feb 19 21:30:23 WARNING[26948]: Unable to get our IP address, Skinny disabled

**Invitado** · 02-19-2005, 09:37 PM

No se escucha en los telefonos SIP los mensajes que se deben

sip debug en la consola de asterisk y busca que error te marca.

**akrall** · 02-19-2005, 09:50 PM

No se escucha en los telefonos SIP los mensajes que se deben

realmente no marca ningun error, solo no se escucha nada...

Aqui esta lo que muestra el log:

Sip read:
INVITE sip:1000@10.0.0.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.1.2;branch=z9hG4bKd1e9ebf0a83bbd42
From: "Anton K" <sip:grandstream1@10.0.0.10;user=phone>;tag=49a864 e9e3d058bc
To: <sip:1000@10.0.0.10;user=phone>
Contact: <sip:grandstream1@10.0.1.2;user=phone>
Supported: replaces
Call-ID: 55ab643a8487a705@10.0.1.2
CSeq: 10224 INVITE
User-Agent: Grandstream HT286 1.0.5.22
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU BSCRIBE
Content-Type: application/sdp
Content-Length: 327

v=0
o=grandstream1 8000 8000 IN IP4 10.0.1.2
s=SIP Call
c=IN IP4 10.0.1.2
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 98
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=ptime:20

13 headers, 16 lines
Using latest request as basis request
Sending to 10.0.1.2 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 98
Peer audio RTP is at port 10.0.1.2:5004
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Found description format iLBC
Capabilities: us - 0x406 (gsm|ulaw|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Found user 'grandstream1'
Looking for 1000 in default
list_route: hop: <sip:grandstream1@10.0.1.2;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.2;branch=z9hG4bKd1e9ebf0a83bbd42
From: "Anton K" <sip:grandstream1@10.0.0.10;user=phone>;tag=49a864 e9e3d058bc
To: <sip:1000@10.0.0.10;user=phone>;tag=as24c7ed5d
Call-ID: 55ab643a8487a705@10.0.1.2
CSeq: 10224 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1000@10.0.0.10>
Content-Length: 0

to 10.0.1.2:5060
-- Executing Goto("SIP/grandstream1-9db6", "default|s|1") in new stack
-- Goto (default,s,1)
-- Executing Wait("SIP/grandstream1-9db6", "1") in new stack
-- Executing Answer("SIP/grandstream1-9db6", "") in new stack
We're at 10.0.0.10 port 11268
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x400 (ilbc)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.2;branch=z9hG4bKd1e9ebf0a83bbd42
From: "Anton K" <sip:grandstream1@10.0.0.10;user=phone>;tag=49a864 e9e3d058bc
To: <sip:1000@10.0.0.10;user=phone>;tag=as24c7ed5d
Call-ID: 55ab643a8487a705@10.0.1.2
CSeq: 10224 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1000@10.0.0.10>
Content-Type: application/sdp
Content-Length: 203

v=0
o=root 26971 26971 IN IP4 10.0.0.10
s=session
c=IN IP4 10.0.0.10
t=0 0
m=audio 11268 RTP/AVP 3 0 98
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 iLBC/8000
a=silenceSupp

ff - - - -

to 10.0.1.2:5060
-- Executing DigitTimeout("SIP/grandstream1-9db6", "5") in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("SIP/grandstream1-9db6", "10") in new stack
-- Set Response Timeout to 10
-- Executing BackGround("SIP/grandstream1-9db6", "demo-congrats") in new stack
-- Playing 'demo-congrats' (language 'en')

Sip read:
ACK sip:1000@10.0.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.2;branch=z9hG4bKd6795570895ba162
From: "Anton K" <sip:grandstream1@10.0.0.10;user=phone>;tag=49a864 e9e3d058bc
To: <sip:1000@10.0.0.10;user=phone>;tag=as24c7ed5d
Contact: <sip:grandstream1@10.0.1.2;user=phone>
Call-ID: 55ab643a8487a705@10.0.1.2
CSeq: 10224 ACK
User-Agent: Grandstream HT286 1.0.5.22
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU BSCRIBE
Content-Length: 0

11 headers, 0 lines

Sip read:

0 headers, 0 lines

Sip read:
BYE sip:1000@10.0.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.2;branch=z9hG4bK97d9fef09928d7f8
From: "Anton K" <sip:grandstream1@10.0.0.10;user=phone>;tag=49a864 e9e3d058bc
To: <sip:1000@10.0.0.10;user=phone>;tag=as24c7ed5d
Call-ID: 55ab643a8487a705@10.0.1.2
CSeq: 10225 BYE
User-Agent: Grandstream HT286 1.0.5.22
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SU BSCRIBE
Content-Length: 0

10 headers, 0 lines
Sending to 10.0.1.2 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.2;branch=z9hG4bK97d9fef09928d7f8
From: "Anton K" <sip:grandstream1@10.0.0.10;user=phone>;tag=49a864 e9e3d058bc
To: <sip:1000@10.0.0.10;user=phone>;tag=as24c7ed5d
Call-ID: 55ab643a8487a705@10.0.1.2
CSeq: 10225 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1000@10.0.0.10>
Content-Length: 0

to 10.0.1.2:5060
== Spawn extension (default, s, 5) exited non-zero on 'SIP/grandstream1-9db6'
Destroying call '55ab643a8487a705@10.0.1.2'

Sip read:

0 headers, 0 lines

**Invitado** · 02-19-2005, 09:52 PM

No se escucha en los telefonos SIP los mensajes que se deben

postea tu sip.conf

**akrall** · 02-19-2005, 09:54 PM

No se escucha en los telefonos SIP los mensajes que se deben

Tampoco la prueba del echo en la extension 600 jala

no me escucho de regreso..

**akrall** · 02-19-2005, 09:55 PM

No se escucha en los telefonos SIP los mensajes que se deben

sip.conf

;
; SIP Configuration example for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/username@domain to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;

[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
; if asterisk was compiled with OSP support.
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet

;pedantic=yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
;tos=184 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;videosupport=yes ; Turn on support for SIP video
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)

disallow=all ; First disallow all codecs
allow=gsm ; Allow codecs in order of preference
allow=ulaw ; Allow codecs in order of preference
allow=ilbc ;
musicclass=default ; Sets the default music on hold class for all SIP calls
; This may also be set for individual users/peers
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since SIP is incapable
; of performing a "hairpin" call.
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)

;compactheaders = yes ; send compact sip headers.

;
; If regcontext is specified, Asterisk will dynamically
; create and destroy a NoOp priority 1 extension for a given
; peer who registers or unregisters with us. The actual extension
; is the 'regexten' parameter of the registering peer or its
; name if 'regexten' is not provided. More than one regexten may be supplied
; if they are separated by '&'. Patterns may be used in regexten.
;
;regcontext=sipregistrations
;
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => user[:secret[:authuser]]@host[

ort][/extension]
;
; If no extension is given, the 's' extension is used. The extension
; needs to be defined in extensions.conf to be able to accept calls
; from this SIP proxy (provider)
;
; host is either a host name defined in DNS or the name of a
; section defined below.
;
; Examples:
;
;register => 1234

assword@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345

assword@sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local
; extension 1234 in extensions.conf default context, unless you define
; unless you configure a [sip_proxy] section below, and configure a context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both directions

;registertimeout=20 ; retry registration calls every 20 seconds (default)
;callevents=no ; generate manager events when sip ua performs events (e.g. hold)

;---------------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk behind
; a NAT device to communicate with services on the outside.

;externip = 200.201.202.203 ; Address that we're going to put in outbound SIP messages
; if we're behind a NAT

; The externip and localnet is used
; when registering and communicating with other proxies
; that we're registered with
;externhost=foo.dyndns.net ; Alternatively you can specify an
; external host, and Asterisk will
; perform DNS queries periodically. Not
; recommended for production
; environments! Use externip instead
;externrefresh=10 ; How often to refresh externhost if
; used
; You may add multiple local networks. A reasonable set of defaults
; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network

; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router).
; If you have one-way audio problems, you usually have problems with your NAT
; configuration or your firewalls support of SIP+RTP ports.
; You configure Asterisk choice of RTP ports for incoming audio in rtp.conf
;
;nat=no ; Global NAT settings (Affects all peers and users)
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581
; never = Never attempt NAT mode or RFC3581 support
; route = Assume NAT, don't send rport
; (work around more UNIDEN bugs)

;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis.
;rtnoupdate=yes ; do not send the update request over realtime.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered when the registration expires
; the friend will vanish from the configuration until requested
; again. If set to an integer, friends expire
; within this number of seconds instead of the
; same as the registration interval

;-----------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; permit permit
; deny deny
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; useclientcode useclientcode
; accountcode accountcode
; setvar setvar
; callerid callerid
; amaflags amaflags
; incominglimit incominglimit
; restrictcid restrictcid
; mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; mask
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout

;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
;type=peer
;context=from-fwd
;host=fwd.pulver.com

;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;usereqphone=yes ; This provider requires ";user=phone" on URI

;------------------------------------------------------------------------------
; Definitions of locally connected SIP phones
;
; type = user a device that calls us
; type = peer a device we place calls to
; type = friend two configurations (peer+user) in one
;
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you propably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open

[grandstream1]
type=friend
context=default
callerid=John Doe <1234> ; Full caller ID, to override the phones config
host=dynamic
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
dtmfmode=rfc8233 ; either RFC2833 or INFO for the BudgeTone
;incominglimit=1 ; permit only 1 outgoing call at a time
; from the phone to asterisk
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
disallow=all ; need to disallow=all before we can use allow=
allow=gsm ; Note: In user sections the order of codecs
allow=ulaw ; Note: In user sections the order of codecs
allow=ilbc ; Pass-thru only unless g729 license obtained

;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;type=friend
;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw

;[snom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blah
;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59 ; IP used until peer registers
;username=snom ; Username to use in INVITE until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;restrictcid=yes ; To have the callerid restriced -> sent as ANI
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!

;[polycom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blahpoly
;host=dynamic ; This peer register with us
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;username=polly ; Username to use in INVITE until peer registers
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don't work properly with "never"

;[pingtel]
;type=friend
;username=pingtel
;secret=blah
;host=dynamic
;insecure=yes ; To match a peer based by IP address only and not peer name
;insecure=very ; To allow registered hosts to call without re-authenticating
;qualify=1000 ; Consider it down if it's 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
;defaultip=192.168.0.60 ; IP address to use if peer has not registred

;[cisco1]
;type=friend
;username=cisco1
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4 ; IP address to use until registration
;username=goran ; Username to use when calling this device before registration

**Invitado** · 02-19-2005, 10:16 PM

No se escucha en los telefonos SIP los mensajes que se deben

ve a http://mike.calle69.net y baja las configuraciones si esas configuraciones no te sirven algo en tu red debe estar bloqueandote los udps para que no pase la voz. El log no marca ningun error de codec ni nada por el estilo no veo que estes omitiendo nada de mucha relevancia baja las configs del sitio que te di aver si mejora. Si no buscale por el lado de tu red.

**akrall** · 02-19-2005, 10:27 PM

No se escucha en los telefonos SIP los mensajes que se deben

ya le di al pedo.... segun los docs de asterisk, tu maquina puede o no tener tarjeta de sonido, y habia leido que iclusive si la tienes la puedes desactivar... pues no!!!!

Resulta que yo la tenia desactivada en el bios.... asi que la active y un reboot... y voila!! ya se escuchan los prompts...

ahora, tnego un pedo con lo tonos, asterisk no quiere tomar los tonos dtfm del telefono, alguna idea?

No se escucha en los telefonos SIP los mensajes que se deben

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